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Avaya Solution & Interoperability Test Lab
Application Notes for Avaya 1100- and 1200-Series IP
Deskphones R3.2 with Avaya Aura™ Communication
Manager R6, Avaya Aura™ Session Manager R6, and
Avaya Modular Messaging R5.2 – Issue 1.0
Abstract
These Application Notes describe a solution comprised of Avaya Aura™ Communication
Manager, Avaya Aura™ Session Manager, Avaya Modular Messaging, and Avaya 1100- and
1200-Series IP Deskphones with SIP software. During testing, the IP Deskphones successfully
registered with Session Manager, placed and received calls to and from SIP and non-SIP
telephones, and executed other telephony features such as conference, transfer, hold, and Off-
PBX-Station (OPS) related features such as Call Pickup, Call Park, Whisper Page, and
Transfer to Voice Mail.
Information in these Application Notes has been obtained through interoperability testing and
additional technical discussions, and was conducted at the Avaya Solution and Interoperability
Test Lab at the request of Avaya 1100- and 1200-Series IP Deskphone Product Management.
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1. Introduction
These Application Notes describe a solution comprised of Avaya Aura™ Communication
Manager, Avaya Aura™ Session Manager, Avaya Modular Messaging, and Avaya 1100- and
1200-Series IP Deskphones with SIP software (formerly known as Nortel 1100- and 1200-Series
SIP Phones). These telephones were originally developed under the Nortel brand, and as such do
not currently support the Avaya Advanced SIP Telephony (AST) protocol implemented in Avaya
9600 Series IP Telephones (SIP). Nevertheless, Communication Manager and Session Manager
have the capability to extend some advanced telephony features to non-AST telephones. The
configuration steps described include how to set up these features as well as the standard calling
features supported by the phones. See Section 4 for a summary of the features supported.
2. Reference Configuration
In the test configuration shown below, the Avaya S8800 Server with Avaya G450 Media
Gateway is configured as an Evolution Server (CM-ES), and supports all of the telephones
shown. The SIP telephone models tested included the 1120E (4 line monochrome), 1140E (6
line monochrome), 1165E (8 line color), 1220 (4 line monochrome), and the 1230 (10 line
monochrome) running SIP firmware. The phones are directly registered to Session Manager and
are supported by Communication Manager configured as an Evolution Server (CM-ES).
Communication between Communication Manager and Avaya Modular Messaging is via
Session Manager, which uses an adaptation module to translate subscriber numbers between the
5-digit extensions used by Communication Manager and the normalized 11-digit numbers used
by Modular Messaging. Modular Messaging supports all telephones for voice messaging.
Figure 1: Sample Configuration
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In general, a SIP telephone originates a call by sending a call request (SIP INVITE message) to
Session Manager, which then routes the call over a SIP trunk to Communication Manager for
origination services. If the call is destined for another local SIP telephone, Communication
Manager routes the call back over the SIP trunk to Session Manager for delivery to the
destination SIP telephone. If the call is destined for an H.323 or Digital telephone, then
Communication Manager terminates the call directly.
These application notes assume that Communication Manager and Session Manager are already
installed and basic configuration steps have been performed. Only steps relevant to SIP
telephone calling features will be described in this document. For further details on configuration
steps not covered in this document, consult the appropriate document in Section 10.
3. Equipment and Software Validated
The following equipment and software/firmware were used for the sample configuration
provided.
Equipment Software/Firmware
Avaya S8800 Server with G450 Media
Gateway
Avaya Aura™ Communication Manager 6.0
Service Pack 0 (Load 345, Update 18246)
Avaya S8800 Server
Avaya Aura™ Session Manager 6.0, Load
600020
Avaya Aura™ System Manager 6.0, Load
600020
Avaya 9630 IP Telephone (SIP) 2.6.0.0
Avaya 9630 IP Telephone (H.323) 3.1.1
Avaya 1603 IP Telephone (SIP) R1.0.0
Avaya 6408D+ Digital Telephone -
Modular Messaging Storage Server 5.2, Service Pack 3 Patch 1
Modular Messaging Application Server 5.2, Service Pack 3 Patch 1
Avaya 1100-series IP deskphones (SIP) 03.02.15.05
Avaya 1200-series IP deskphones (SIP) 03.02.15.05
Table 1: Equipment and Software/Firmware
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4. Calling Features
4.1. Overview
Table 2 below shows the calling features successfully tested. Notes on specific feature
operations are included in Section 4.2. In addition to basic calling capabilities, the Internet
Engineering Task Force (IETF) has defined a supplementary set of calling features in RFC 5359
[13], previously referred to as the SIPPING features. This provides a useful framework to
describe product capabilities and compare features supported by various equipment vendors.
Communication Manager can support many of these features if the telephone can not locally
support them. In addition, features beyond those specified in RFC 5359 can be extended to the
telephone using Communication Manager configured as an Evolution Server.
SUPPORTED FEATURES COMMENTS
Basic Calling features
Extension to extension call
Basic call to non-SIP phones
Intercept tones/displays Reorder with message
Mute
Redial
Call Waiting
Do Not Disturb Section 4.2.1
Speed Dial buttons
Redial from call logs
Compressed codecs G.729A, G.729AB, G.722-64k
Message Waiting Support
SIPPING (RFC 5359) Features
Call Hold
Consultation Hold
Music on Hold
Unattended Transfer
Attended Transfer
Call Forward Unconditional Sections 4.2.2, 5.8
Call Forward Busy Via FNE (Section 5.8)
Call Forward No Answer Via FNE (Section 5.8)
3-way conference - 3rd party added
3-way conference - 3rd party joins
Find-Me Modular Messaging “Find Me” feature
Incoming Call Screening Via Class Of Restriction (Section 5.9)
Outgoing Call Screening Via Class Of Restriction (Section 5.9)
Call Park/Unpark Via FNE (Section 5.8)
Call Pickup Via FNE (Section 5.8)
Additional Station-Side Features
Calling Name/Number Block Via FNE (Section 4.2.3, 5.8, 7.4)
Directed Call Pick-Up Via FNE (Section 5.8)
Priority Call Via FNE (Sections 4.2.4, 5.8, 5.9)
Transfer to Voice Mail Via FNE (Section 5.8)
Whisper Page Via FNE (Section 5.8)
Table 2: SIP Telephony Feature Support
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Some supported features shown in Table 2 can be invoked by dialing a Feature Name Extension
(FNE). Or, a speed dial button on the telephone can be programmed to an FNE. Communication
Manager automatically handles many other standard features such as call coverage, trunk
selection using Automatic Alternate Routing (AAR) and Automatic Route Selection (ARS),
Class Of Service/Class Of Restriction (COS/COR), and voice messaging. Details on operation
and administration for Communication Manager can be found in References [4-6].
4.2. Operational Notes
4.2.1. Do Not Disturb
When Do Not Disturb is activated, the call is not presented to or displayed on the phone. The
call follows the coverage path configured for the extension in Communication Manager. This
feature is locally supported by the telephone, and is recommended instead of the Communication
Manager FNEs for Send All Calls and Send All Calls Cancel.
4.2.2. Call Forward Unconditional
It is recommended that this feature be administered as a Communication Manager FNE rather
than using the local call forward of the telephone. The user of local call forward will not benefit
from any of the call coverage features available in Communication Manager, including coverage
to voice messaging.
4.2.3. Calling Name/Number Block
The Avaya 1100- and 1200-Series IP Deskphones support privacy by means of the Remote-
Party-ID SIP header in the INVITE message. Since Communication Manager supports the
newer Privacy header along with P-Asserted-Identity header, this local feature is not supported.
It is recommended that the Calling Number Block FNE in Communication Manager be used
instead. This can be configured as speed dial button on the telephone (see Sections 5.8 and 7.4).
4.2.4. Priority Call
The telephone may originate priority calls based on the class of service administered for it (see
Section 5.9) or if the user dials the appropriate FNE. Note however, that it will not indicate a
received priority call. Avaya 9600 Series IP Telephones (SIP, H.323) and Avaya 6408D+
Digital Telephones will properly indicate them via distinctive ringing and calling party display.
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5. Configure Avaya Aura™ Communication Manager
This section describes a procedure for setting up a SIP trunk between Communication Manager
serving as an Evolution Server and Session Manager. This includes steps for setting up IP
codecs, an IP network region, SIP signaling group, SIP trunk group, dial plan, class of service,
class of restriction, and call routing. Also, a procedure is described here to configure SIP
telephones and features available with OPS in Communication Manager. Configuration in the
following sections focuses on the fields where a value needs to be entered or modified. Default
values are used for all other fields.
These steps are performed from the Communication Manager System Access Terminal (SAT)
interface. Avaya and third party SIP telephones are configured as Off-PBX Stations (OPS) in
Communication Manager. Communication Manager does not directly control an OPS endpoint,
but its features and calling privileges can be applied to it by associating a local extension with
the OPS endpoint. Similarly, a SIP telephone in Session Manager is associated with an extension
on Communication Manager. SIP telephones register with Session Manager and use
Communication Manager for call origination and termination services, including Feature Name
Extension (FNE) support. Enter the save translation command after completing this section.
5.1. Capacity Verification
Before a SIP trunk or OPS endpoints can be configured, it is necessary to verify if there is
enough capacity.
Step Description
1.
Enter the display system-parameters customer-options command. Verify that there are
sufficient Maximum Off-PBX Telephones – OPS licenses. If not, contact an authorized
Avaya account representative to obtain additional licenses.
display system-parameters customer-options Page 1 of 11
OPTIONAL FEATURES
G3 Version: V16 Software Package: Enterprise
Location: A System ID (SID): 1
Platform: 28 Module ID (MID): 1
USED
Platform Maximum Ports: 65000 296
Maximum Stations: 36000 124
Maximum XMOBILE Stations: 41000 0
Maximum Off-PBX Telephones - EC500: 36000 1
Maximum Off-PBX Telephones - OPS: 36000 101
Maximum Off-PBX Telephones - PBFMC: 36000 0
Maximum Off-PBX Telephones - PVFMC: 36000 0
Maximum Off-PBX Telephones - SCCAN: 0 0
Maximum Survivable Processors: 313 0
(NOTE: You must logoff & login to effect the permission changes.)display system-
parameters customer-options
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2.
Proceed to Page 2 of the OPTIONAL FEATURES form. Verify that the number of
Maximum Administered SIP Trunks supported by the system is sufficient for the
number of SIP trunks needed. If not, contact an authorized Avaya account representative
to obtain additional licenses.
Note: Each SIP call between two SIP endpoints requires two SIP trunks for the duration
of the call. The license file installed on the system controls the maximum permitted.
display syste
m
-parameters customer
-
o
ptions Page 2 of 11
OPTIONAL FEATURES
IP PORT CAPACITIES USED
Maximum Administered H.323 Trunks: 12000 0
Maximum Concurrently Registered IP Stations: 18000 11
Maximum Administered Remote Office Trunks: 12000 0
Maximum Concurrently Registered Remote Office Stations: 18000 0
Maximum Concurrently Registered IP eCons: 414 0
Max Concur Registered Unauthenticated H.323 Stations: 100 0
Maximum Video Capable Stations: 18000 0
Maximum Video Capable IP Softphones: 18000 0
Maximum Administered SIP Trunks: 24000 172
Maximum Administered Ad-hoc Video Conferencing Ports: 24000 0
Maximum Number of DS1 Boards with Echo Cancellation: 522 0
Maximum TN2501 VAL Boards: 128 0
Maximum Media Gateway VAL Sources: 250 1
Maximum TN2602 Boards with 80 VoIP Channels: 128 0
Maximum TN2602 Boards with 320 VoIP Channels: 128 0
Maximum Number of Expanded Meet-me Conference Ports: 300 0
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5.2. IP Codec Set
This section describes the steps for administering an IP codec set, which is used in the IP
network region when Communication Manager communicates with the SIP telephones via
Session Manager.
Step Description
1.
Enter the change ip-codec-set n command, where n is a number between 1 and 7,
inclusive. IP codec sets are used in Section 5.3 for configuring an IP network region to
specify which codec sets may be used within and between network regions. For the
compliance testing, G.722-64K, G.711MU, G.729A, and G.729AB were tested. If only
one codec should be used, then only specify the one that is to be used.
Note: for G.729 interoperability between Avaya 1100- and 1200-Series IP Deskphones
and Avaya 9600 Series SIP Telephones, the configuration file settings for all telephones
should match that in the IP codec set. For the sample configuration shown below, the
9600 SIP Telephone configuration file should include the line: SET ENABLE_G729 “2”,
and the 1100- and 1200-Series IP Deskphone configuration files should include the line
“G729_ENABLE_ANNEXB YES”. See References [7, 8] and Section 7.2.
change ip-codec-set 1 Page 1 of 2
IP Codec Set
Codec Set: 1
Audio Silence Frames Packet
Codec Suppression Per Pkt Size(ms)
1: G.722-64K 2 20
2: G.711MU n 2 20
3: G.729AB n 2 20
4:
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5.3. IP Network Region
This section describes the steps for administering an IP network region, which is used when
Communication Manager communicates with the SIP telephones via Session Manager.
Step Description
1.
Enter the change ip-network-region n command, where n is a number between 1 and
250 inclusive and configure the following as shown in the display screen below:
Authoritative Domain – Set to avaya.com for the sample configuration. This
should match the SIP Domain value configured in Session Manager.
Intra-region IP-IP Direct Audio – Set to yes to allow direct IP-to-IP audio
connectivity between endpoints registered to Communication Manager or Session
Manager in the same IP network region.
Inter-region IP-IP Direct Audio – Set to yes to allow direct IP-to-IP audio
connectivity between endpoints registered to Communication Manager or Session
Manager in different IP network regions.
Codec Set – Set the codec set number as provisioned in Section 5.2.
change ip-network-region 1 Page 1 of 20
IP NETWORK REGION
Region: 1
Location: Authoritative Domain: avaya.com
Name: CM and SIP Phones
MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes
Codec Set: 1 Inter-region IP-IP Direct Audio: yes
UDP Port Min: 2048 IP Audio Hairpinning? y
UDP Port Max: 65535
DIFFSERV/TOS PARAMETERS
Call Control PHB Value: 0
Audio PHB Value: 46
Video PHB Value: 26
802.1P/Q PARAMETERS
Call Control 802.1p Priority: 6
Audio 802.1p Priority: 6
Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS
5.4. IP Node Names
This section describes the steps for administering a node name for Session Manager to be used in
the configuration of the SIP signaling group.
Step Description
1.
Use the change node-names ip command to add a new node name for Session Manager.
change node-names ip Page 1 of 2
IP NODE NAMES
Name IP Address
SM1 10.1.2.70
default 0.0.0.0
procr 10.1.2.90
procr6 ::
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5.5. SIP Signaling Group
This section describes the steps for administering a signaling group for communication between
Communication Manager and Session Manager.
Step Description
1.
Enter the command add signaling-group n, where n is an available signaling group and
configure the following as shown in the display screen below:
Group Type – Set to sip.
Transport Method – Set to tls.
IMS Enabled – Set to n.
Near-end Node Name - Set to procr.
Near-end Listen Port - Defaults to 5061 for TLS.
Far-end Node Name - Set to the node name configured in Section 5.4.
Far-end Listen Port - Defaults to 5061 for TLS.
Far-end Network Region - Set to the Region configured in Section 5.3.
Far-end Domain - Set to avaya.com for the sample configuration. This should
match the SIP Domain value configured in Session Manager.
Direct IP-IP Audio Connections – Set to y.
ad
d
signaling-group 60 Page 1 of 1
SIGNALING GROUP
Group Number: 60 Group Type: sip
IMS Enabled? n Transport Method: tls
Q-SIP? n SIP Enabled LSP? n
IP Video? n Enforce SIPS URI for SRTP? y
Peer Detection Enabled? y Peer Server:
Near-end Node Name: procr Far-end Node Name: SM1
Near-end Listen Port: 5061 Far-end Listen Port: 5061
Far-end Network Region: 1
Far-end Domain: avaya.com
Bypass If IP Threshold Exceeded? n
Incoming Dialog Loopbacks: eliminate RFC 3389 Comfort Noise? n
DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y
Session Establishment Timer(min): 3 IP Audio Hairpinning? n
Enable Layer 3 Test? n Initial IP-IP Direct Media? n
H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 10
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5.6. SIP Trunk Group
This section describes the steps for administering a trunk group for communication between
Communication Manager and Session Manager.
Step Description
1.
Issue the command add trunk-group n, where n is an unallocated trunk group and
configure the following as shown in the display screen below:
Group Type – Set to the Group Type field to sip.
Group Name – Enter any descriptive name.
TAC (Trunk Access Code) – Set to any available trunk access code.
Service Type – Set to tie.
Signaling Group – Set to the Group Number field value configured in Section
5.5. (i.e., 60)
Number of Members – Allowed values are between 0 and 255. Set to a value
large enough to accommodate the number of SIP telephone extensions being used.
Note: Each SIP call between two SIP endpoints (whether internal or external) requires
two SIP trunk members for the duration of the call. The license file installed on the system
controls the maximum permitted.
add trunk-group 60 Page 1 of 21
TRUNK GROUP
Group Number: 60 Group Type: sip CDR Reports: y
Group Name: SM1 COR: 1 TN: 1 TAC: 160
Direction: two-way Outgoing Display? n
Dial Access? n Night Service:
Queue Length: 0
Service Type: tie Auth Code? n
Member Assignment Method: auto
Signaling Group: 60
Number of Members: 120
2.
Proceed to Page 3 and set Numbering Format to private.
change trunk-group 60 Page 3 of 21
TRUNK FEATURES
ACA Assignment? n Measured: none
Maintenance Tests? y
Numbering Format: private
UUI Treatment: service-provider
Replace Restricted Numbers? n
Replace Unavailable Numbers? N
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5.7. Define System Features
This section describes the steps for administering system wide call features and options related to
OPS in Communication Manager.
Step Description
1.
Use the change system-parameters features command and navigate to Page 18 to
administer system wide features for the SIP telephones. Those related to features listed in
Table 2 are shown outlined in red. These are all standard Communication Manager
features.
change syste
m
-parameters features Page 18 of 19
FEATURE-RELATED SYSTEM PARAMETERS
INTERCEPT TREATMENT PARAMETERS
Invalid Number Dialed Intercept Treatment: tone
Invalid Number Dialed Display:
Restricted Number Dialed Intercept Treatment: tone
Restricted Number Dialed Display:
Intercept Treatment On Failed Trunk Transfers? n
WHISPER PAGE
Whisper Page Tone Given To: paged
change system-parameters features Page 19 of 19
FEATURE-RELATED SYSTEM PARAMETERS
IP PARAMETERS
Direct IP-IP Audio Connections? y
IP Audio Hairpinning? y
Synchronization over IP? n
CALL PICKUP
Maximum Number of Digits for Directed Group Call Pickup: 4
Call Pickup on Intercom Calls? y Call Pickup Alerting? n
Temporary Bridged Appearance on Call Pickup? y Directed Call Pickup? y
Extended Group Call Pickup: simple
Enhanced Call Pickup Alerting? N
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5.8. Define the Dial Plan
This section describes the steps for administering the dial plan, including overall dial plan
format, Feature Access Codes (FACs), and Feature Name Extensions (FNEs).
Step Description
1.
Use the change dialplan analysis command to define the dial plan formats used in the
system. This includes all telephone extensions, Feature Name Extensions (FNEs), and
Feature Access Codes (FACs). To define the FNEs for the features listed in Table 2, a
Feature Access Code (FAC) must also be specified for the corresponding feature. In the
sample configuration, telephone extensions are five digits long and begin with 3, FNEs
are five digits beginning with 7, and the FACs have formats as indicated with Call Type
“fac”. Note: a FAC of “8” was used for AAR routing by a voice mail hunt group, the
configuration for which is not included in these Application Notes. See Reference [11]
for more information.
change dialplan analysis Page 1 of 12
DIAL PLAN ANALYSIS TABLE
Location: all Percent Full: 2
Dialed Total Call Dialed Total Call Dialed Total Call
String Length Type String Length Type String Length Type
0 3 fac
1 3 dac
2 5 ext
3 5 ext
4 5 ext
5 5 ext
6 3 fac
7 5 ext
8 1 fac
9 1 fac
* 2 fac
# 2 fac
2.
Use the change private-numbering command to add an entry as shown below for the
calling extensions that will be using the trunk to Session Manager. The entry specifies the
format that the calling number will have in outgoing calls. Set Ext Len to the length of
the calling extensions, Ext Code to an initial set of digits that covers the extension range,
Trk Grp(s) to the trunk group number defined in Section 5.6, and Total Len to the
length of the calling extensions. In the sample configuration, the extension is sent
unchanged. Note: if the Trk Grp(s) field is left blank, as in the first entry, this formatting
will be applied to all trunk groups in the system.
change private-numbering 0 Page 1 of 2
NUMBERING - PRIVATE FORMAT
Ext Ext Trk Private Total
Len Code Grp(s) Prefix Len
5 2 5 Total Administered: 3
5 3 60 5 Maximum Entries: 540
5 4 60 5
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3.
Use change feature-access-codes to define the access codes for the FNEs highlighted in
red. The following screens have been abbreviated to highlight those FACs involved in
supporting the FNEs and the AAR FAC.
change feature-access-codes Page 1 of 10
FEATURE ACCESS CODE (FAC)
Answer Back Access Code: 625
Attendant Access Code:
Auto Alternate Routing (AAR) Access Code: 8
Auto Route Selection (ARS) - Access Code 1: 9 Access Code 2:
Automatic Callback Activation: *5 Deactivation: #5
Call Forwarding Activation Busy/DA: *2 All: 612 Deactivation: #2
Call Forwarding Enhanced Status: Act: Deactivation:
Call Park Access Code: 624
Call Pickup Access Code: *6
CAS Remote Hold/Answer Hold-Unhold Access Code: #6
change feature-access-codes Page 2 of 10
FEATURE ACCESS CODE (FAC)
Contact Closure Pulse Code:
Directed Call Pickup Access Code: 654
Directed Group Call Pickup Access Code:
Emergency Access to Attendant Access Code:
EC500 Self-Administration Access Codes:
Enhanced EC500 Activation: 660 Deactivation: 661
Enterprise Mobility User Activation: Deactivation:
Extended Call Fwd Activate Busy D/A All: Deactivation:
Extended Group Call Pickup Access Code: 641
change feature-access-codes Page 3 of 10
FEATURE ACCESS CODE (FAC)
PASTE (Display PBX data on Phone) Access Code:
Personal Station Access (PSA) Associate Code: Dissociate Code:
Per Call CPN Blocking Code Access Code: 615
Per Call CPN Unblocking Code Access Code: 616
Posted Messages Activation: Deactivation:
Priority Calling Access Code: *7
Program Access Code: *0
Refresh Terminal Parameters Access Code: 694
Remote Send All Calls Activation: Deactivation:
Self Station Display Activation:
Send All Calls Activation: *3 Deactivation: #3
Station Firmware Download Access Code:
change feature-access-codes Page 4 of 10
FEATURE ACCESS CODE (FAC)
Transfer to Voice Mail Access Code: #9
Trunk Answer Any Station Access Code:
User Control Restrict Activation: 691 Deactivation: 692
Voice Coverage Message Retrieval Access Code:
Voice Principal Message Retrieval Access Code:
Whisper Page Activation Access Code: 620
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4.
FNEs are defined using the change off-pbx-telephone feature-name-extensions set n
command, where n is a number between 1 and 99 and will default to 1 if n is not
specified. This command is used to support both SIP telephones and Extension to
Cellular. The highlighted fields correspond to those features listed as supported in Table
2. The fields that have been left blank correspond to those more appropriate for Extension
to Cellular.
change off-pbx-telephone feature-name-extensions set 1 Page 1 of 2
EXTENSIONS TO CALL WHICH ACTIVATE FEATURES BY NAME
Set Name:
Active Appearance Select: 70030
Automatic Call Back: 70003
Automatic Call-Back Cancel: 70004
Call Forward All: 70005
Call Forward Busy/No Answer: 70006
Call Forward Cancel: 70007
Call Park: 70008
Call Park Answer Back: 70009
Call Pick-Up: 70010
Calling Number Block: 70012
Calling Number Unblock: 70013
Conditional Call Extend Enable:
Conditional Call Extend Disable:
Conference Complete:
Conference on Answer: 70011
Directed Call Pick-Up: 70014
Drop Last Added Party: 70015
change off-pbx-telephone feature-name-extensions set 1 Page 2 of 2
EXTENSIONS TO CALL WHICH ACTIVATE FEATURES BY NAME
Exclusion (Toggle On/Off): 70016
Extended Group Call Pickup: 70025
Held Appearance Select: 70017
Idle Appearance Select:
Last Number Dialed: 70019
Malicious Call Trace: 70029
Malicious Call Trace Cancel: 70021
Off-Pbx Call Enable: 70027
Off-Pbx Call Disable: 70028
Priority Call: 70000
Recall:
Send All Calls: 70001
Send All Calls Cancel: 70002
Transfer Complete:
Transfer On Hang-Up: 70022
Transfer to Voice Mail: 70023
Whisper Page Activation: 70026
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5.9. Specify Class of Service (COS) and Class Of Restriction (COR)
This section describes the steps for administering the COS and COR, which affects what calling
features and feature options are permitted for defined groups of telephone users.
Step Description
1.
Use the change cos-group n command, where n is a class of service group number, to set
the appropriate service permissions to support the corresponding features (shown in
outlined in red). For the sample configuration, COS group 1 was used. On Page 2, set
the value of VIP Caller to “y” only if all calls made by telephones with this COS should
be priority calls. Note that priority can be requested on a call-by-call basis by using the
Priority Call FNE (see Section 5.8). Priority call indication (e.g., distinctive ring and
display of “Priority”) is only supported on Avaya Digital and 9600 Series IP telephones
(H.323 and SIP).
change cos-group 1 Page 1 of 2
CLASS OF SERVICE COS Group: 1 COS Name:
0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Auto Callback n y y n y n y n y n y n y n y n
Call Fwd-All Calls n y y y y n n y y n n y y n n n
Data Privacy n n n y n y y y y n n n n y y y
Priority Calling n y n n n n n n n y y y y y y n
Console Permissions y y y n n n n n n n n n n n n n
Off-hook Alert n n n n n n n n n n n n n n n n
Client Room n n n n n n n n n n n n n n n n
Restrict Call Fwd-Off Net n n y y y y y y y y y y y y y y
Call Forwarding Busy/DA n y n n n n n n n n n n n n n n
Personal Station Access (PSA) n n n n n n n n n n n n n n n n
Extended Forwarding All n y n n n n n n n n n n n n n n
Extended Forwarding B/DA n y n n n n n n n n n n n n n n
Trk-to-Trk Transfer Override n y n n n n n n n n n n n n n n
QSIG Call Offer Originations n n n n n n n n n n n n n n n n
Contact Closure Activation n n n n n n n n n n n n n n n n
change cos-group 1 Page 2 of 2
CLASS OF SERVICE
0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
VIP Caller n n n n n n n n n n n n n n n n
Masking CPN/Name Override n n n n n n n n n n n n n n n n
Call Forwarding Enhanced y y y y y y y y y y y y y y y y
Priority Ip Video n n n n n n n n n n n n n n n n
Ad-hoc Video Conferencing n n n n n n n n n n n n n n n n
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2.
Use the change cor n command, where n is a class of restriction number, to enable
applicable calling features. To use the Directed Call Pickup feature, the Can Use
Directed Call Pickup and Can Be Picked Up By Directed Call Pickup fields must be
set to “y” for the affected stations. In the sample configuration, the telephones were
assigned to COR 1. Note that Page 4 can be used to implement a form of centralized call
screening for groups of stations and trunks.
change cor 1 Page 1 of 23
CLASS OF RESTRICTION
COR Number: 1
COR Description: Trunk
FRL: 0 APLT? y
Can Be Service Observed? y Calling Party Restriction: none
Can Be A Service Observer? y Called Party Restriction: none
Time of Day Chart: 1 Forced Entry of Account Codes? n
Priority Queuing? n Direct Agent Calling? y
Restriction Override: none Facility Access Trunk Test? n
Restricted Call List? n Can Change Coverage? n
Access to MCT? y Fully Restricted Service? n
Group II Category For MFC: 7 Hear VDN of Origin Annc.? n
Send ANI for MFE? n Add/Remove Agent Skills? n
MF ANI Prefix: Automatic Charge Display? n
Hear System Music on Hold? y PASTE (Display PBX Data on Phone)? n
Can Be Picked Up By Directed Call Pickup? y
Can Use Directed Call Pickup? y
Group Controlled Restriction: inactive
change cor 1 Page 4 of 23
CLASS OF RESTRICTION
CALLING PERMISSION (Enter "y" to grant permission to call specified COR)
0? y 15? y 30? y 44? y 58? y 72? y 86? y
1? y 16? y 31? y 45? y 59? y 73? y 87? y
2? y 17? y 32? y 46? y 60? y 74? y 88? y
3? y 18? y 33? y 47? y 61? y 75? y 89? y
4? y 19? y 34? y 48? y 62? y 76? y 90? y
5? y 20? y 35? y 49? y 63? y 77? y 91? y
6? y 21? y 36? y 50? y 64? y 78? y 92? y
7? y 22? y 37? y 51? y 65? y 79? y 93? y
8? y 23? y 38? y 52? y 66? y 80? y 94? y
9? y 24? y 39? y 53? y 67? y 81? y 95? y
10? n 25? y 40? y 54? y 68? y 82? y 96? y
11? y 26? y 41? y 55? y 69? y 83? y 97? y
12? y 27? y 42? y 56? y 70? y 84? y 98? y
13? y 28? y 43? y 57? y 71? y 85? y 99? y
FS; Reviewed:
SPOC 09/07/2010
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©2010 Avaya Inc. All Rights Reserved.
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5.10. SIP Stations
This section describes the steps for administering OPS stations in Communication Manager and
associating the OPS station extensions with the telephone numbers of the Avaya 1100-and 1200-
Series IP Deskphones. The configuration is the same for all phones except for the desired
number of call appearances as detailed in Step 3. Note that the corresponding users must be
configured in Session Manager. There are two methods to sequence these steps:
1. Configure the station and off-PBX-station forms for each user in Communication
Manager. Then configure the corresponding user in Session Manager, being sure to
check the “Use Existing Stations” box (see Section 6).
2. Configure the user in Session Manager, being sure to leave the “Use Existing Stations”
box unchecked (see Section 6). Session Manager will automatically generate the
corresponding station and off-PBX-station information in Communication Manager.
Then use the change station command in Communication Manager to add other
configuration data, such as Coverage Path, MWI Served User Type, and additional call
appearances, if needed.
Method 2 was used in the sample configuration. For method 1, perform the following steps for
each user; then follow the steps in Section 6. For method 2, follow the steps in Section 6 first;
then use change station n to modify any station parameters as described below using the station
form in this section as a guide.
FS; Reviewed:
SPOC 09/07/2010
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©2010 Avaya Inc. All Rights Reserved.
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Step Description
1.
Enter the add station n command, where n is an available extension in the dial plan, to
administer an OPS station. On Page 1 of the form configure the following fields as shown
in the display screen below:
Type – Set to 9630SIP.
Port – Leave blank. (Once the form is submitted, a virtual port is assigned, e.g.,
S00022)
Name – Enter any descriptive name.
Coverage Path – Enter the coverage path number defined for this telephone (e.g.,
for coverage to voice mail).
add station 30043 Page 1 of 6
STATION
Extension: 30043 Lock Messages? n BCC: 0
Type: 9630SIP Security Code: TN: 1
Port: Coverage Path 1: 60 COR: 1
Name: 1165E Coverage Path 2: COS: 1
Hunt-to Station:
STATION OPTIONS
Time of Day Lock Table:
Loss Group: 19
Message Lamp Ext: 30043
Display Language: english Button Modules: 0
Survivable COR: internal
Survivable Trunk Dest? y IP SoftPhone? n
IP Video? N
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Solution & Interoperability Test Lab Application Notes
©2010 Avaya Inc. All Rights Reserved.
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2.
Proceed to Page 2 of the form. Set MWI Served User Type to sip-adjunct.
add station 30043 Page 2 of 6
STATION
FEATURE OPTIONS
LWC Reception: spe
LWC Activation? y Coverage Msg Retrieval? y
Auto Answer: none
CDR Privacy? n Data Restriction? n
Idle Appearance Preference? n
Per Button Ring Control? n Bridged Idle Line Preference? n
Bridged Call Alerting? n
Active Station Ringing: single
H.320 Conversion? n Per Station CPN - Send Calling Number?
EC500 State: enabled
MWI Served User Type: sip-adjunct
Coverage After Forwarding? s
Direct IP-IP Audio Connections? y
Emergency Location Ext: 30043 Always Use? n IP Audio Hairpinning? N
3.
Proceed to Page 4 of the form and add the desired number of call-appr entries in the
BUTTON ASSIGNMENTS section. This governs how many concurrent calls can be
supported. Avaya 1100-Series IP Deskphones have the capability of handling 11 call
appearances (10 for the 1200-Series), but display only one local call appearance button
when idle (see display in Section 7.4 Step 3). So the number of entries shown below are
not required to match that displayed on the telephone. Three are configured here to
support conferencing scenarios.
add station 30043 Page 4 of 6
STATION
SITE DATA
Room: Headset? n
Jack: Speaker? n
Cable: Mounting: d
Floor: Cord Length: 0
Building: Set Color:
ABBREVIATED DIALING
List1: List2: List3:
BUTTON ASSIGNMENTS
1: call-appr 5:
2: call-appr 6:
3: call-appr 7:
4: 8:
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